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VoIP & SIP
Troubleshooting Guides

Practical guides for engineers covering SIP response codes, call quality, protocol internals, infrastructure, platform debugging, UCaaS, and compliance. All free. If you have a trace to analyze, the SIP trace analyzer does the interpretation for you.

83

Guides

9

Categories

Free

Always

SIP Response Codes

16 guides

SIP 302 Moved Temporarily

Call forwarding, redirect handling, loop prevention

SIP 401 Unauthorized

Digest authentication flow, realm mismatch, credential errors

SIP 403 Forbidden

IP not authorized, account suspended, ACL blocks

SIP 404 Not Found

User or domain does not exist on this server

SIP 407 Proxy Authentication

Proxy auth, how it differs from 401, fixing auth loops

SIP 408 Request Timeout

No response, firewall blocking, wrong port, DNS failure

SIP 480 Temporarily Unavailable

Extension not registered, DND, registration expired

SIP 481 Call Leg Not Found

Dialog state lost, server restart, B2BUA issues

SIP 486 Busy Here

Endpoint on another call, call waiting, overflow

SIP 488 Not Acceptable Here

Codec mismatch, SRTP vs RTP, incompatible SDP

SIP 500 Internal Server Error

PBX crash, database failure, misconfiguration

SIP 503 Service Unavailable

Server overload, upstream gateway failure, capacity limits

SIP Registration Failed

401/403/404 errors, DNS failures, firewall blocks

SIP 487 Request Terminated

CANCEL races, forking branch losses, proxy timeouts

SIP 491 Request Pending

Re-INVITE glare, RFC 3261 retry timer, SBC interference

SIP 603 Decline

Global failure semantics, 603 vs 486 vs 487, forking

Call Quality and Media

12 guides

One Way Audio in VoIP Calls

RTP one direction, NAT, firewall, SDP address, SRTP mismatch

VoIP Packet Loss

Acceptable thresholds, causes, QoS fixes

VoIP Jitter Troubleshooting

Acceptable levels, causes, jitter buffer tuning

What is good MOS, how to measure, how to improve

VoIP Calls Drop After 30 Seconds

SIP Timer H, NAT binding expiry, session timer failures

DTMF Not Working in VoIP

RFC 2833, in-band, SIP INFO transport methods

VoIP Bandwidth Requirements

Bandwidth per codec, concurrent call planning, WAN sizing

RTP Timeout Mid-Call

NAT pinhole expiry, asymmetric routing, SIP vs RTP timeout

No Ringback Tone in VoIP

Local vs remote ringback, 180/183 SDP, early media silence

Echo on VoIP Calls

Acoustic, hybrid, and network echo, AEC failures

VoIP Codec Comparison

Opus, G.722, G.729, G.711 — bandwidth, quality, support

Transcoding in VoIP

When transcoding happens, CPU and quality costs, avoidance

SIP Protocol

8 guides

SIP Dialog States Explained

Early, confirmed, terminated states and forking edge cases

SIP ACK and the Three-Way Handshake

End-to-end vs hop-by-hop ACK, 30-second silent drop

Re-INVITE vs UPDATE

RFC 3311 differences, when each is correct, interop

Early Media and 183 Session Progress

100rel, PRACK, why early media often fails

B2BUA vs SIP Proxy

Architectural differences, Call-ID continuity, identification

Record-Route and Route Headers

Route set construction, missing Record-Route symptoms

Via Header Chains and Loops

Branch parameters, loop detection, rport and received

SIP Codec Negotiation and SDP Offer-Answer

RFC 3264 mechanics, payload types, fmtp parameters

Infrastructure

14 guides

VoIP Firewall Configuration

Ports to open, disable SIP ALG, firewall rules, NAT

VoIP NAT Traversal

STUN, TURN, ICE, SBC, symmetric NAT solutions

VoIP QoS Configuration

DSCP EF marking, priority queuing, VLAN separation

SIP Load Balancing

DNS SRV, dispatcher, round-robin, health checks

SIP OPTIONS Ping

Keepalive, trunk health monitoring, failure detection

SIP Trunking Explained

How trunks work, registration vs static, common issues

SIP Trunking vs PRI

Technical differences, cost comparison, migration steps

SIP Trunk Failover Configuration

DNS SRV failover, Asterisk/FreeSWITCH, SBC redundancy

VoIP Toll Fraud Prevention

Brute force detection, fail2ban, routing controls

VoIP Monitoring Tools

SNMP, RTCP, Homer, CDR analysis, alerting

VoIP E911 Compliance

Dispatchable location, RAY BAUM Act, Kari's Law

Ghost Calls and Phantom Rings

Scanner traffic, firewall blocking, ALG and PBX defenses

SIP Registration Loops and Flapping

Expires negotiation, NAT keepalive, Contact rewrites

Symmetric RTP and Latching

Paired NAT traversal techniques, failure modes

Security and Compliance

9 guides

VoIP Voice Encryption TLS and SRTP

TLS signaling, SRTP media, SDES vs DTLS-SRTP

SIP TLS Certificate Setup

Certificate requirements, Let's Encrypt for SIP

STIR/SHAKEN Explained

Call authentication, attestation A/B/C, PASSporT tokens

VoIP HIPAA Compliance

TLS/SRTP requirements, BAAs, call recording, audit controls

VoIP PCI-DSS Compliance

DTMF masking, call recording, network segmentation

VoIP SOC 2 Compliance

Trust Services Criteria, audit evidence, encryption

TRACED Act and Robocall Mitigation Database

RMD registration, FCC requirements, traceback obligations

GDPR for VoIP and Call Recording

Legal basis, retention, DSARs, breach notification

The verstat Header Explained

Reading STIR/SHAKEN verification results, attestation levels

Platform Guides

13 guides

Asterisk SIP Debugging

Debug logging, CLI commands, chan_sip vs PJSIP

Asterisk PJSIP Configuration

pjsip.conf setup, transport, TLS, NAT, trunks

Asterisk Call Quality Troubleshooting

RTP statistics, jitter buffers, codec issues

Kamailio SIP Troubleshooting

Debug logging, xlog, routing failures, database

Kamailio Dispatcher Load Balancing

Dispatcher module, routing algorithms, failover

Kamailio NAT Traversal with RTPEngine

RTPEngine setup, SDP rewriting, ICE and SRTP

FreeSWITCH SIP Troubleshooting

Sofia SIP logging, fs_cli, registration, audio

OpenSIPS Troubleshooting

Debug logging, MI commands, routing script

3CX SIP Troubleshooting

Trunk registration, one-way audio, packet capture

FusionPBX Troubleshooting

SIP debug, registration failures, dialplan issues

Grandstream SIP Configuration

UCM setup, endpoint provisioning, SIP trunk config

RingCentral SIP Troubleshooting

BYOC, call quality, registration, Direct Routing

Cisco CUCM SIP Trunk

CUCM config, security profiles, codec regions, RTMT

UCaaS and Cloud

6 guides

UCaaS SIP Troubleshooting

Teams, Zoom Phone, RingCentral, call failures

UCaaS Call Quality Troubleshooting

Fix choppy, robotic, one-way audio on cloud systems

Microsoft Teams SIP Trunk

Operator Connect vs Direct Routing, SBC, TLS

Teams Direct Routing SIP Issues

TLS certificates, SBC config, codec mismatches

Zoom Phone SIP Trunk BYOC

BYOC config, SBC setup, E.164 format, common issues

Twilio SIP Trunk Configuration

Elastic SIP trunking, credential lists, IP ACL

WebRTC

2 guides

WebRTC to SIP Gateway Issues

ICE failures, DTLS-SRTP negotiation, codec mismatches

WebRTC SIP Interoperability

Protocol differences, gateway patterns, media bridging

AI and VoIP

3 guides

AI Voice Agent SIP Integration

Architecture, codecs, latency optimization, echo

SIP Trunking for AI Agents

Carrier selection, G.711, concurrency, latency

VoIP API Integration Guide

SIPSymposium API, programmatic analysis, JSON output

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